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Pass the Cisco CCNP Collaboration 350-801 Questions and answers with Dumpstech

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Viewing questions 81-90 out of questions

Questions # 81:

A customer is deploying a SIP IOS gateway for a customer who requires that in-band DTMF relay is first priority and out-of-band DTMF relay is second priority. Which 10$ entry sets the required priority?

Options:

dtmf-relay cisco-rtp

dtmf-relay sip-kpml cisco-rtp

sip-notify dtmf-relay rtp-nte

dtmf-relay rtp-nte sip-notify

Answer

Questions # 82:

An engineer must configure a SIP integration between cisco UCM and cisco Unity Connection. These components are configured already:

Question # 82

Drag and drop the components from the left into the order they must be configured on the right.

Question # 82

Options:

Answer

Answer:

Explanation

Summary of Steps:

Create a SIP Trunk Security Profile.

Create a SIP Profile.

Create the SIP Trunk in Cisco UCM.

Configure Route Patterns for routing calls to Unity Connection.

Configure SIP ports in Cisco Unity Connection.

Set the Voicemail Pilot Number in Cisco UCM.

Configure the Voicemail Ports in Cisco UCM.

Assign Voicemail Profile to the appropriate users.

Questions # 83:

Refer to the exhibit.

https://i.postimg.cc/C57TkczG/image.png

Question # 83

A call is falling to establish between two SIP Devices The called device answers with these SOP Which SOP parameter causes issue?

Options:

The calling device did not offer a ptime value

The media stream is set to send only

The payload for G.711ulaw must be 18.

The RTP port is set to 0.

Answer

Explanation

The RTP port is used to send and receive media packets during a call. If the RTP port is set to 0, the called device will not be able to send or receive media packets, and the call will fail.

The other options are not correct because:

A. The calling device did not offer a ptime value: The ptime value is used to specify the amount of time between each media packet. If the calling device does not offer a ptime value, the called device will use the default value of 20 milliseconds.

B. The media stream is set to sendonly: The media stream is set to sendonly when the called device is only able to send media packets, and not receive them. This is not a problem, and the call will still succeed.

C. The payload for G.711ulaw must be 18: The payload for G.711ulaw is the type of media packet that is used. The payload must be set to 18 for G.711ulaw, but this is not a problem, and the call will still succeed.

Questions # 84:

Which call routing pattern is used for phone numbers that are in the E.164 format?

Options:

/+.! Route Pattern

\+.! Route pattern

\+.! Translation Pattern

\+1.[2-9]XX[2-9]XXXXXXX called Party Transformation Pattern

Answer

Questions # 85:

During the Cisco IP Phone registration process the TFTP download fails. What are two reasons for this issue? (Choose two)

Options:

The DNS server was not specified, which is needed to resolve the DHCP server IP address.

Option 100 string was not specified, or an incorrect Option 100 string was specified.

The Cisco IP Phone does not know the IP address of the TFTP server.

The Cisco IP Phone does not know the IP address of any of the Cisco UCM Subscriber nodes.

Option 150 string was not specified, or an incorrect Option 150 string was specified.

Answer

C, E

Questions # 86:

Which traversal client must be deployed in an enterprise network when Mobile and Remote Access is configured?

Options:

Cisco Unified Border Element

Cisco Expressway-C

Cisco Expressway-E

Cisco IM and Presence

Answer

Questions # 87:

Which two protocols can be configured for the Cisco Unity Connection and Cisco UCM integration? (Choose two.)

Options:

H. 323

SIP

SCCP

MGCP

RTP

Answer

B, C

Explanation

The two protocols that can be configured for the Cisco Unity Connection and Cisco UCM integration are SIP and SCCP.

SIP, or Session Initiation Protocol, is a signaling protocol used for initiating, maintaining, and terminating real-time sessions, including voice, video, and messaging applications.

SCCP, or Skinny Client Control Protocol, is a Cisco proprietary signaling protocol used for controlling Cisco IP phones.

H.323 is an older signaling protocol that is no longer widely used. MGCP, or Media Gateway Control Protocol, is a protocol used for controlling media gateways. RTP, or Real-time Transport Protocol, is a protocol used for transporting real-time data, such as voice and video

Questions # 88:

An engineer configures a Cisco Unified Border Element and must ensure that the codecs negotiated meet the ITSP requirements. The ITSP supports G.7llulaw and G.729 for audio and H.264 for video. The preferred voice codec is G.711. Which configuration meets this requirement?

Options:

Answer

Questions # 89:

Refer to the exhibit A call from +1 613 555 1234 that is sent out through the Nice Gateways should result in a calling party of 001 613 555 1234 with the numbering type "international' Which configuration accomplishes this goal?

Question # 89

Options:

\+.001! pre-dot 1 international

\+1.1 none pre-dot 001 international

\+.! pre-dot 00 international

613XXXXXXX none +011 internationa

Answer

Questions # 90:

Which information is needed to restore the backup of a Cisco UCM publisher successfully?

Options:

the TFTP server details

the application credentials for Cisco UCM

the security password for Cisco UCM

the FTP server details

Answer

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Viewing questions 81-90 out of questions

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